On Sat, Jun 15, 2013 at 04:24:21PM -0400, Andre Goree wrote: > Thanks so much for your suggestions. > > I'm running 1.0.x (yes, archaic, and in fact my actual task is > migrating this system to asterisk11+Freepbx -- very fun in and of > itself without regards to this issue...but I digress), and so I needed > to run "pri debug span <span>", which I've now done. I attempted the > call again have pasted the debug output here: > http://pastebin.com/cHHnMfh6
You mentioned the telco receiving a DISCONNECT almost immediatly. Your debug is only up to a PROGRESS. I only have experience with euroisdn but callflow would be: ->SETUP <-CALLPROCEDING <-PROGRESS <-CONNECT ->CONNECT ACK ->DISCONNECT (eg from caller) <-RELEASE ->RELEASE COMPLETE But PROGRESS means the recipient is generating some audio (your unreachable message?). If this is an error message you would expect a RELASE from the telco after the recording if the caller doesn't hangup first. You should study the difference of zap->zap and sip->zap callsetup. -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
